Network & Firewall Reference¶
A quick reference for network requirements. For symptom-based help, see Troubleshooting.
Ports¶
| Traffic | Protocol / ports | Notes |
|---|---|---|
| RTP (call audio) | UDP 16384–32768 | Must be open in both directions |
| SIP signalling | Proxy port shown in the Partner Portal (remote/VPS deployments) | On-premise deployments use DNS SRV with port 0 — see SIP Registration |
Router / firewall checklist¶
- Disable SIP ALG on the router — it is the most common cause of no-audio, one-way audio, and dropped calls.
- Allow inbound and outbound UDP on the SIP and RTP ports (asymmetric rules cause one-way audio).
- Avoid double NAT for phones; use bridge mode or port forwarding on both layers if unavoidable.
- Enable QoS to prioritise voice traffic; use a dedicated voice VLAN where possible.
- Prefer wired connections over Wi-Fi for desk phones.
Bandwidth and quality targets¶
| Metric | Target |
|---|---|
| Bandwidth per concurrent call (G.711) | ~100 Kbps up and down |
| Packet loss | Below 1% |
| Latency | Under 150 ms |
| Jitter | Below 30 ms |
NAT keep-alive¶
Phones behind NAT should have NAT keep-alive enabled (typically a 30-second interval) to prevent lost registrations and mid-call drops.